Integrating multimedia capabilities with circuit-switched calls

ABSTRACT

The present invention monitors call signaling events stemming from a circuit-switched call between a caller and a called party and controls a packet-session between user agents on respective endpoints associated with the caller and called party. The endpoints may include any type of computational device capable of facilitating the packet-session over a packet-switched network. Control of the user agents may be provided via a proxy for the user agents and may use the session initiation protocol (SIP), or like session control protocol for communications.

This application is a Continuation of U.S. patent application Ser. No.09/960,554, entitled INTEGRATING MULTIMEDIA CAPABILITIES WITHCIRCUIT-SWITCHED CALLS, filed Sep. 21, 2001, currently pending, thedisclosure of which is hereby incorporated by reference in its entirety.

This application also claims the benefit of Provisional Application Ser.No. 60/308,177, filed Jul. 27, 2001, the disclosure of which isincorporated herein by reference.

FIELD OF THE INVENTION

The present invention relates to multimedia communications, and inparticular, relates to integrating multimedia capabilities withcircuit-switched calls.

BACKGROUND OF THE INVENTION

The acceptance of network applications and the Internet has given risefor a need to associate voice communications with various network-basedapplications and functions. While engaged in a telephone conference,users often share applications, such as whiteboarding applications, andweb pages to enhance communications. In most instances, users initiatethe voice call and then establish an application sharing arrangementindependently of the voice call.

Numerous software packages have attempted to integrate voice and datacommunications over packet-switched networks. Unfortunately, theavailability of quality packet-based voice systems is low while thecircuit-switched voice systems are widely available. As such, attemptshave been made to associate circuit-switched voice calls withpacket-switched multimedia applications.

Previous attempts to integrate circuit switched voice calls andmultimedia sessions have required proprietary protocols or cumbersomeprotocols, such as H.323, The lack of flexibility and complexity ofthese protocols have suppressed their acceptance and availability.Accordingly, there is a need for an efficient and easy to implementtechnique for integrating circuit-switched voice calls andpacket-switched multimedia capability. Further, there is a need toprovide such integration without requiring proprietary protocols and byusing accepted standards that are readily available.

SUMMARY OF THE INVENTION

The present invention monitors call signaling events stemming from acircuit-switched call between a caller and a called party and controls apacket-session between user agents on respective endpoints associatedwith the caller and called party. The endpoints may include any type ofcomputational device capable of facilitating the packet-session over apacket-switched network. Control of the user agents may be provided viaa proxy for the user agents and may use the session initiation protocol(SIP), or like session control protocol for communications.

Directory numbers for the circuit-switched, customer premise equipmentsupporting the circuit-switched call may be associated withcommunication addresses for the endpoints. During operation, thedirectory numbers for the calling and called party are used to identifythe addresses of the respective endpoints. When a circuit-switched callis initiated or established, a call signaling trigger is identified, andthe packet session is established between the user agents of theendpoints. Upon completion of the circuit-switched call, a correspondingcall signaling trigger is identified, and the packet session is ended.

A dedicated integration service may be used to facilitate interactionwith the circuit switched network to identify call-signaling events andcooperate with a proxy to control the packet session between the useragents. Alternatively, the integration service may be combined with theproxy alone, or with other network devices.

Those skilled in the art will appreciate the scope of the presentinvention and realize additional aspects thereof after reading thefollowing detailed description of the preferred embodiments inassociation with the accompanying drawing figures.

BRIEF DESCRIPTION OF THE DRAWING FIGURES

The accompanying drawing figures incorporated in and forming a part ofthis specification illustrate several aspects of the invention, andtogether with the description serve to explain the principles of theinvention.

FIG. 1 is an illustration of a communication environment according toone embodiment of the present invention.

FIG. 2 is a block representation of a SIP integration server accordingto one embodiment of the present invention.

FIGS. 3A and 3B are an exemplary communication flow according oneembodiment of the present invention.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

The embodiments set forth below represent the necessary information toenable those skilled in the art to practice the invention and illustratethe best mode of practicing the invention. Upon reading the followingdescription in light of the accompanying drawing figures, those skilledin the art will understand the concepts of the invention and willrecognize applications of these concepts not particularly addressedherein. It should be understood that these concepts and applicationsfall within the scope of the disclosure and the accompanying claims.

With reference to FIG. 1, an exemplary communication environment 10capable of carrying out the concepts of the present invention isillustrated. The communication environment 10 is depicted as including asystem signaling 7 (SS7) network and a session initiation protocol (SIP)enabled packet switched network 14. The SIP enabled network 14 mayinclude any type of packet-switched network having devices using SIP tofacilitate communications between two or more devices.

The SS7 network 12 is an advanced intelligent network (AIN) capable ofproviding call signaling for voice-based communications over the publicswitched telephone network (PSTN) and well known services such astoll-free dialing, automatic redial, call back, and calling numberdelivery (caller ID). The primary components of a typical SS7 network 12include service switching points (SSP) 16, service transfer points (STP)18, and service control points (SCP) 20.

A typical SSP 16, such as Nortel Networks Limited's DMS100, providesstandard voice switching and is equipped with SS7 hardware, software,and signaling links. The SSPs 16 are the “end-points” of an SS7 networkand typically reside between the customer premise equipment 22 and theSS7 network 12. SSPs 16 also typically provide the end-to-end circuitfor the voice transport and perform most of the basic call processingrequired to setup and terminate voice calls.

The STPs 18 are typically reliable, high-speed packet switches thatroute SS7 signaling messages throughout the SS7 network 12. The SCPs 20are computing platforms that run applications providing enhanced servicelogic to default call processing actions of the SSPs 16, depending onthe application. Those skilled in the art will have an appreciation andunderstanding of the SS7 network 12 and like signaling networks.

Using AIN concepts, the SSPs 16 will temporarily halt call processingfor certain events or points during a call and send information relatingto the events to the SCP 20. The SCP 20 will either continue callprocessing or perform an action, such as release, redirect, or terminatethe call. As such, the SCP 20 is routinely engaged in call processingand has information pertaining to the call being process and eventstaking place during such processing. The present invention provides amechanism for the SIP network 14 to interact with the SCP 20 to identifyevents during call processing as well as influence call processing asdesired.

The Internet Engineering Task Force's RFC 2543, which is incorporated inits entirety by reference, provides the ability to establish sessionsbetween endpoints 24 over a packet switched network, such as the SIPnetwork 14, running the internet protocol (IP). Once established, thesesessions can exchange media capabilities and set up multiple media pathsbetween the endpoints 24 based on their capabilities. A SIP endpoint 24will support a User Agent (UA).

User Agents register their ability to receive calls with a SIP proxyserver 26 by sending “REGISTER” messages to the SIP proxy server 26. The“REGISTER” message informs the SIP proxy server 26 of the SIP uniformresource locator (URL), which identifies the User Agent to the network.The “REGISTER” message also contains information about how to reach thespecific User Agent over the SIP network 14. For instance, the“REGISTER” message may provide the User Agent's IP address and port inwhich the User Agent will monitor or facilitate communications.

Typically, when a User Agent wants to initiate a call to another UserAgent, it will send an “INVITE” message to the SIP proxy server 26specifying the targeted User Agent in the “TO” header. Identification ofa User Agent takes the form of a SIP URL, <username>@<domain>, such asjanedoe@nortelnetworks.com. The SIP proxy server 26 will use the SIP URLin the “TO” header of the message to determine if the User Agent isregistered. The username is usually unique within the namespace of thespecified domain.

If the targeted User Agent has registered, the SIP proxy server 26 willforward the “INVITE” message directly to the targeted User Agent. TheUser Agent responds with a 200 OK message to the originating User Agentvia the SIP proxy server 26, and a session between the two User Agentswill be established as per the message exchange required in the SIPspecification. Capabilities are passed between the two User Agents asparameters embedded within the session setup messages such as INVITE,200 OK, and ACK. Media capabilities can also be exchanged using the SIP“INFO” message. Capabilities are typically described using the SessionDescription Protocol (SDP). Once User Agents are in an active sessionwith each other and understand each other's capabilities, the respectiveendpoints 24 providing the User Agents can exchange the specified mediacontent.

Reference is made to the Internet Engineering Task Force draft“draft-rosenberg-sip-3pcc-00.ext” for SIP third party call control,which is outlined below.

Third party call control may take the following form. A centralcontroller first calls one of the participants, A, and presents theINVITE message without any media. When this call is complete, thecontroller has the SDP needed to communicate with A. The controller thenuses the SDP to initiate a call to participant B. When this call iscompleted, the controller has the SDP needed to communicate with B. Thisinformation is then passed to A. The result is that there is a call legbetween the controller and A, a call leg between the controller and B,but media between A and B.

In the preferred embodiment of the invention, an application server isconfigured to interact with the SS7 network 12 or like circuit switchednetwork via the SCP 20. The application server is generally referred toherein as a “SIP Integration Server” or SIS 28, which monitors callevents from the SS7 network 12 and performs third party call controlbetween the Calling and Called User Agents on the SIP network based onthose events. The SIS 28 may be affiliated with a database 30 to provideinformation for operation and pertaining to the directory numbers,addresses and the like for User Agents and CPEs 22.

As shown in FIG. 2, the SIS 28 may be a typical web server having acentral processing unit (CPU) 38 with the requisite memory 40 containingthe software and data necessary for operation. The CPU 38 is associatedwith a network interface 44 facilitating communications with otherdevices, such as the endpoints 24, SIP proxy server 26, SCP 20, and thedatabase 30 through any number of local area networks, routers, switchesand hubs in traditional fashion.

Subscribers to the SIS 28 will have certain AIN triggers provisionedagainst their directory number for the CPE 22 within their local SSP 16.The triggers correspond to events during call processing. Whenever thesubscriber is involved with a telephone call, the SSP 16 will route theappropriate triggers to the SIS service for interpretation. It should benoted that this disclosure is not tied to the mechanism by which AINmessages are routed to the SIS 28. In addition, SIS subscribers willhave User Agents installed on their personal computer or other computingdevice that has access to the SIP network 14. These User Agents can beany SIP enabled applications that know how to register with the SIPproxy server 26, can establish SIP sessions, and can exchangemulti-media content of one or more types. For another example of a SIPuser agent benefiting from the present invention, please refer to U.S.patent application Ser. No. 09/666,583, filed Sep. 21. 2001, entitled“AUTOMATED WEB BROWSER SYNCHRONIZATION,” the disclosure of which isincorporated herein by reference.

SIP User Agents register with the SIP proxy server 26 as specified inthe SIP specification and preferably identify their username in the SIPURL to be equivalent to the directory number that is assigned to theirtelephone. For example if John Smith has a directory number of (555)991-1234, then John's SIP User Agent would register with the username of<5559911234>@<domain> with the SIP proxy server 26. Such registration ispreferred because the directory numbers are the main identifier usedwithin the AIN triggers received by the SIS 28 from the SSP 16.

When the SIS 28 needs to interact with the SIP network 14, it will sendSIP messages to the SIP proxy server 26 and identify the intendedrecipient of the message using the corresponding directory number. TheSIP proxy server 26 will forward the message to the appropriate UserAgent. Notably, this disclosure depicts the SIP Integration Server andthe SIP Proxy as two separate entities; however, the functionality ofboth may reside within the same platform or even within the sameapplication. Thus, the concepts of the present invention may beimplemented with the proxy functionality embedded within the SIS 28, andvice versa.

A SIP session is initiated when the SU receives a trigger from the SCP20 that the voice call for its user has been answered. On receiving thistrigger from the SCP 20, the SIS 28 first attempts to establish a SIPsession with the caller's User Agent without specifying any mediacapabilities. The caller's User Agent accepts the session invitation andwill respond with its capability descriptions. The SIS 28 then attemptsto establish a session with the called user's User Agent and sends thecapabilities of the caller's User Agent. The called User Agent acceptsthe session invitation and responds with its own capabilitydescriptions. The SIS 28 then forwards the called User Agents capabilitydescriptions back to the caller's User Agent to complete the capabilitynegotiations, and the two User Agents begin exchanging content on theirestablished media path(s).

The SIP Session is terminated whenever the SIS 28 receives a Disconnecttrigger from the SCP 20 indicating that either side has released thevoice call. Once this notification is received, the SIS 28 sends a SIPBYE message to both User Agents and the SIP Session is terminated.

Turning now to FIGS. 3A and 3B, an exemplary communication flow isdescribed for setting up and releasing calls using the SIS 28 tosynchronize SIP multi-media sessions with voice calls. Those skilled inthe art will recognize the exemplary communication flow highlights onlysome of the applications made possible by the present invention. Theconcepts and architecture of the present invention allow for numerousservice extensions, which should become apparent to those skilled in theart upon reading this disclosure.

For the exemplary flow of FIGS. 3A and 3B, User A places a call to UserB and a SIP Session is established with multi-media capability, such aswhiteboarding or application sharing. Assume User A has a telephonenumber of (555) 444-1111 and a SIP User Agent A application running onan associated personal computer (endpoint 24) that supportswhiteboarding. Further assume that User Agent A has previouslyregistered with the SIP proxy server 26 as5554441111@<service_provider_domain>. User B has a telephone number of(555) 333-2222 and a SIP User Agent B application running on his/herpersonal computer that also supports whiteboarding. The User Agent forUser B has previously registered with the SIP proxy server 26 as555333222@<service_provider_domain>.

Initally, User A uses his CPE 22 (i.e., telephone) to place a call toUser B by dialing 333-2222. User A's SSP 16 sends an InfoAnalyzed AINtrigger to the SCP 20 servicing the SIS 28. The InfoAnalyzed triggercontains both the directory number of the calling user (User A) and thedirectory number of the called user (User B). The SCP 20 forwards theInfoAnalyzed trigger to the SIS 28 (step 100). The SIS 28 looks up thesubscriber profile for User A on database 30 by indexing on the callinguser's (User A's) directory number. The SIS 28 finds the profile forUser A and maintains the knowledge that User A is placing a call to UserB.

The SIS 28 replies to the SCP with a Continue response thus allowing theSSP 16 to continue to setup the call to User B's CPE 22 (step 102). TheSSP 16 servicing User B's CPE 22 sends a TerminationAttempt trigger tothe SCP 20, which forwards it on to the SIS 28 (step 104). The SIS 28looks up the subscriber profile for User B by indexing on the calleduser's (User B) directory number. The SIS 28 finds the profile for UserB and maintains the knowledge that User A is placing a call to User B.

The SIS 28 replies by sending an AuthorizeTermination response back tothe SSP 16 via the SCP 20 (step 106). The SSP 16 terminates the call toUser B's CPE 22, which begins to ring. User B answers the call bylifting the handset of the respective CPE 22. The SSP 16 servicing UserB's CPE 22 sends an Answer trigger to the SIS 28 via the SCP 20 (step108). The SIS 28 now knows that the voice call between User A and User Bhas been answered.

To set up an associated SIP session, the SIS 28 establishes a sessionwith the caller's (User A) User Agent first. To do this, the SIS 28sends an “INVITE” message to the SIP proxy server 26 with the usernamein the TO: field set to the telephone number of UserA—(5554441111@<service_provider_domain>) (step 110). This initial INVITEmessage typically does not contain any capability information. The SIPproxy server 26 forwards the message on to the User Agent on endpoint 24for User A (step 112).

User A's User Agent replies with a 200 OK message and specifies it'scapability information, such as media type and coding/decoding (CODEC)support, in the message body (step 114). The SIP proxy server 26forwards the 200 OK back to the SIS 28 (step 116).

The SIS 28 sends an “INVITE” message to the SIP proxy server 26 with theusername of the TO: field set to the telephone number of User(555333222@<service_provider_domain>) (step 118). Also included in thismessage is the capability description received in the 200 OK messagefrom the caller's (User A) User Agent. The SIP proxy server 26 forwardsthe message on to the User Agent on endpoint 24 for User B (step 120).

The User Agent for User B replies with a 200 OK message and specifiesits capability information in the message body (step 122). The SIP proxyserver 26 forwards the 200 OK response back to the SIS 28 (step 124).The SIS 28 acknowledges the 200 OK response from the User Agent of UserB by sending an ACK message to User Agent for User B via the SIP proxyserver 26 (steps 126 and 128).

The SIS 28 now needs to send the capability information it received fromUser B's User Agent to User A's User Agent. Accordingly, the SIS 28builds an ACK message with the capability description received from the200 OK message from User Agent of User B and sends the ACK message toUser A's User Agent via the SIP proxy server 26 (steps 130 and 132).Media capability information could also have been sent in a second“INVITE” message as opposed to the ACK as described in association withthe Third Party Call Control IETF draft discussed above.

A SIP session is now set up between the User Agents for User A and UserB, thus allowing users to perform the multimedia function(s) asdescribed within the session description of the message body. Wheneither user ends the voice call by hanging up their handset of the CPE22, the SCP 20 will send an appropriate Disconnect trigger to the SIS 28(step 136). The SIS 28 will terminate the SIP Session by sending “BYE”messages to each User Agent via the SIP proxy server 26 (steps 138through 144).

Those skilled in the art will recognize improvements and modificationsto the preferred embodiments of the present invention. All suchimprovements and modifications are considered within the scope of theconcepts disclosed herein and the claims that follow.

1. A method for integrating separate circuit-switched voice andpacket-switched data sessions, comprising: a. receiving notification ofat least one call signaling event for a separate circuit-switched callbetween a caller and a called party over a circuit-switched network; andb. controlling a separate packet-switched session for a packet-sessionbetween a caller user agent for a caller endpoint and a called partyuser agent on a called party endpoint based on at least one of the callsignaling events for the separate circuit-switched call, wherein thecaller and called party are associated with the caller and called partyendpoints, such that the caller and called party may conduct concurrentvoice and data sessions.
 2. The method of claim 1, wherein thecontrolling step comprises communicating with the caller and calledparty user agents via at least one proxy for at least one of the calleruser agent and called party user agent.
 3. The method of claim 1,wherein the caller and the called party are registered with at least oneproxy using directory numbers associated with the caller and the calledparty respectively.
 4. The method of claim 1, wherein the controllingstep is effected using session initiation protocol (SIP).
 5. The methodof claim 1, wherein the packet-session is a SIP session.
 6. The methodof claim 1, wherein the receiving notification step comprises monitoringtriggers corresponding to the call signaling events provided by a callsignaling control system in the circuit-switched network.
 7. The methodof claim 1, further comprising: a. identifying directory numbers for thecaller and called party; and b. determining addresses of the caller andcalled party user agents to use for the packet-session based on thedirectory numbers.
 8. The method of claim 7, wherein the addresses forthe caller and called party user agents include uniform resourcelocators corresponding to the directory numbers for the caller andcalled party.
 9. The method of claim 1, wherein: a. the receivingnotification step comprises receiving notification of a call signalingtrigger representing initiation or establishment of the circuit-switchedcall between the caller and the called party; and b. the controllingstep comprises initiating the packet-session between the caller useragent and the called party user agent upon identifying the callsignaling trigger representing the initiation or establishment of thecircuit-switched call between the caller and the called party.
 10. Themethod of claim 1 wherein: a. the receiving notification step furthercomprises receiving notification of a call signaling triggerrepresenting completion of the circuit-switched call between the callerand the called party; and b. the controlling step comprises ending thepacket-session between the caller user agent and the called party useragent upon identifying the call signaling trigger representingcompletion of the circuit-switched call between the caller and thecalled party.
 11. The method of claim 1 wherein communications with thecaller and called party user agents are facilitated using sessioninitiation protocol (SIP) via a SIP proxy for the caller and calledparty user agents and: a. the receiving notification step comprises: i.receiving notification of a call signaling trigger representinginitiation or establishment of the circuit-switched call between thecaller and the called party; and ii. receiving notification of a callsignaling trigger representing completion of the circuit-switched callbetween the caller and the called party; and b. the controlling stepcomprises: i. initiating the packet-session between the caller useragent and the called party user agent upon identifying the callsignaling trigger representing initiation or establishment of thecircuit-switched call between the caller and the called party; and ii.ending the packet-session between the caller user agent and the calledparty user agent upon identifying the call signaling triggerrepresenting completion of the circuit-switched call between the callerand the called party.
 12. A SIP integration server, comprising: at leastone processor; at least one memory device storing instructionsexecutable by the at least one processor to: a. receive notification ofat least one call signaling event for a separate circuit-switched callbetween a caller and a called party over a circuit-switched network; andb. control a separate packet-switched session for a packet-sessionbetween a caller user agent for a caller endpoint and a called partyuser agent on a called party endpoint based on at least one of the callsignaling events for the separate circuit-switched call, wherein thecaller and called party are associated with the caller and called partyendpoints, such that the caller and called party may conduct concurrentvoice and data sessions.
 13. The SIP integration server of claim 12,wherein the instructions executable to control a separatepacket-switched session comprises instructions executable to communicatewith the caller and called party user agents via at least one proxy forat least one of the caller user agent and called party user agent. 14.The SIP integration server of claim 12, wherein the caller and thecalled party are registered with at least one proxy using directorynumbers associated with the caller and the called party respectively.15. The SIP integration server of claim 12, wherein the instructionsexecutable to control a separate packet-switched session use sessioninitiation protocol (SIP).
 16. The SIP integration server of claim 12,wherein the packet-session is a SIP session.
 17. The SIP integrationserver of claim 12, wherein the instructions executable to receivenotification step comprises instructions executable to monitor triggerscorresponding to the call signaling events provided by a call signalingcontrol system in the circuit-switched network.
 18. The SIP integrationserver of claim 12, further comprising: a. instructions executable toidentify directory numbers for the caller and called party; and b.instructions executable to determine addresses of the caller and calledparty user agents to use for the packet-session based on the directorynumbers.
 19. The SIP integration server of claim 18, wherein theaddresses for the caller and called party user agents include uniformresource locators corresponding to the directory numbers for the callerand called party.
 20. The SIP integration server of claim 12, wherein:a. the instructions executable to receive notification step compriseinstructions executable to receive notification of a call signalingtrigger representing initiation or establishment of the circuit-switchedcall between the caller and the called party; and b. the instructionsexecutable to control a separate packet-switched session compriseinstructions executable to initiate the packet-session between thecaller user agent and the called party user agent upon identifying thecall signaling trigger representing the initiation or establishment ofthe circuit-switched call between the caller and the called party. 21.The SIP integration server of claim 12, wherein: a. the instructionsexecutable to receive notification further comprise instructionsexecutable to receive notification of a call signaling triggerrepresenting completion of the circuit-switched call between the callerand the called party; and b. the instructions executable to control aseparate packet-switched session comprise instructions executable to endthe packet-session between the caller user agent and the called partyuser agent upon identifying the call signaling trigger representingcompletion of the circuit-switched call between the caller and thecalled party.
 22. The SIP integration server of claim 12, whereincommunications with the caller and called party user agents arefacilitated using session initiation protocol (SIP) via a SIP proxy forthe caller and called party user agents and: a. the instructionsexecutable to receive notification comprise: i. instructions executableto receive notification of a call signaling trigger representinginitiation or establishment of the circuit-switched call between thecaller and the called party; and ii. instructions executable to receivenotification of a call signaling trigger representing completion of thecircuit-switched call between the caller and the called party; and b.the instructions executable to control a separate packet-switchedsession comprise instructions executable to: i. initiate thepacket-session between the caller user agent and the called party useragent upon identifying the call signaling trigger representinginitiation or establishment of the circuit-switched call between thecaller and the called party; and ii. end the packet-session between thecaller user agent and the called party user agent upon identifying thecall signaling trigger representing completion of the circuit-switchedcall between the caller and the called party.
 23. A system forintegrating separate circuit-switched voice and packet-switched datasessions, comprising: a signaling control point operable to transmit atrigger when a calling user connected to a called user by acircuit-switched call handled by the signaling control point indicatesinterest in establishing a packet-switched data session with the calleduser; and a SIP integration server operable, in response to the trigger,to exchange messages with user agents of the calling user and the calleduser to establish the packet-switched data session.
 24. The system ofclaim 23, wherein the SIP integration server is operable to exchangemessages with the user agents which specify capabilities of the useragents to establish the packet-switched data session.
 25. The system ofclaim 24, wherein the SIP integration server is operable to exchangemessages with the user agents by: sending at least one invite message tothe user agent of the called user; receiving at least one responsemessage from the user agent of the called user, the response messagespecifying capabilities of the user agent of the called user; andsending at least one message to the user agent of the calling userspecifying capabilities of the user agent of the called user.
 26. Thesystem of claim 23, wherein: the signaling control point is operable tosend a message to the SIP integration server when either the callinguser or the called user terminates the circuit-switched call; and theSIP integration server is operable to terminate the packet-switched datasession between the calling user and the called user on receipt of themessage indicating termination of the circuit-switched call.
 27. Thesystem of claim 23, further comprising at least one SIP Proxy forcoupling between the SIP integration server and the user agents, the atleast one proxy mediating communications between the SIP integrationserver and the user agents.
 28. The system of claim 27, wherein the atleast one SIP Proxy and the SIP integration server are integrated in acommon server.
 29. A method for integrating separate circuit-switchedvoice and packet-switched data sessions, the method comprising:operating a signaling control point to transmit a trigger when a callinguser connected to a called user by a circuit-switched call handled bythe signaling control point indicates interest in establishing apacket-switched data session with the called user; and operating a SIPintegration server, in response to the trigger, to exchange messageswith user agents of the calling user and the called user to establishthe packet-switched data session.
 30. The method of claim 29, comprisingoperating the SIP integration server to exchange messages with the useragents which specify capabilities of the user agents to establish thepacket-switched data session.
 31. The method of claim 30, comprisingoperating the SIP integration server to exchange messages with the useragents by: sending at least one invite message to the user agent of thecalled user; receiving at least one response message from the user agentof the called user, the response message specifying capabilities of theuser agent of the called user; and sending at least one message to theuser agent of the calling user specifying capabilities of the user agentof the called user.
 32. The method of claim 29, comprising: operatingthe signaling control point to send a message to the SIP integrationserver when either the calling user or the called user terminates thecircuit-switched call; and operating the SIP integration server toterminate the packet-switched data session between the calling user andthe called user on receipt of the message indicating termination of thecircuit-switched call.
 33. The method of claim 29, further comprisingcoupling at least one SIP Proxy between the SIP integration server andthe user agents, and operating the at least one proxy to mediatecommunications between the SIP integration server and the user agents.34. The method of claim 33, wherein the at least one SIP Proxy and theSIP integration server are integrated in a common server.